This is the NEWS for chan_ss7. This version is maintained by Anders Baekgaard (ab@dicea.dk).
Please send bug reports, feature requests etc. to chan_ss7@dicea.dk.

New in version 1.0.94-beta
- Fixed buffer overflows in config.c.
- Fixed loss of IDLE CICs.
- Fixed segmentation fault when using "combined" attribute for linksets.
- Fixed block/unblock of last cic not possible bug.
- Fixed handling of dial request supporting multiple audio formats.
- Support for STP signalling, see file ss7.conf.template.single-link for config.
- Jitter buffer handling (thanks to Martin Vít, sponsored by www.voipex.cz).
- H324M support (thanks to Klaus Darilion).
- Fixed a bug that could cause one-way audio in some cases where DTMF codes are sent.
- Fixed a bug where receive fifo is no longer being read.
- New configuration parameters for link, rxgain and txgain, specifies gain values.
- New configuration parameter for link, relaxdtmf, specifies whether to use relax dtmf.
- Fixed handling of timeout after received suspend message.
- Handling of Chinese SS7 variant: new variant config parameter for linksets (SS7 or CHINA)
  (Thanks to lin.miao@openvox.cn).

New in verion 1.0.0
- Compatible with asterisk 1.2.x and 1.4.x.
- MTP stack placed in standalone executable.
- New loadshare config parameter for linksets (None, linkset, combined).
- New combined config parameter for linksets. Linksets having the same combined
  setting and having loadshare=combined share signalling channels.
- New auto_block config parameter for links. When set to yes, the CICs on that link
  are blocked when signalling on the link is lost.
- The schannel entry in link description in configuration file may specify remote MTP stack.
- PDU dump is now in PCAP format, suitable for wireshark.
- Lots and lots of clean ups and fixes.